FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavcodec/speexdec.c
Date: 2025-04-25 22:50:00
Exec Total Coverage
Lines: 532 774 68.7%
Functions: 22 31 71.0%
Branches: 236 454 52.0%

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1 /*
2 * Copyright 2002-2008 Xiph.org Foundation
3 * Copyright 2002-2008 Jean-Marc Valin
4 * Copyright 2005-2007 Analog Devices Inc.
5 * Copyright 2005-2008 Commonwealth Scientific and Industrial Research Organisation (CSIRO)
6 * Copyright 1993, 2002, 2006 David Rowe
7 * Copyright 2003 EpicGames
8 * Copyright 1992-1994 Jutta Degener, Carsten Bormann
9
10 * Redistribution and use in source and binary forms, with or without
11 * modification, are permitted provided that the following conditions
12 * are met:
13
14 * - Redistributions of source code must retain the above copyright
15 * notice, this list of conditions and the following disclaimer.
16
17 * - Redistributions in binary form must reproduce the above copyright
18 * notice, this list of conditions and the following disclaimer in the
19 * documentation and/or other materials provided with the distribution.
20
21 * - Neither the name of the Xiph.org Foundation nor the names of its
22 * contributors may be used to endorse or promote products derived from
23 * this software without specific prior written permission.
24
25 * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
26 * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
27 * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
28 * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
29 * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
30 * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
31 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
32 * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
33 * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
34 * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
35 * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
36 *
37 * This file is part of FFmpeg.
38 *
39 * FFmpeg is free software; you can redistribute it and/or
40 * modify it under the terms of the GNU Lesser General Public
41 * License as published by the Free Software Foundation; either
42 * version 2.1 of the License, or (at your option) any later version.
43 *
44 * FFmpeg is distributed in the hope that it will be useful,
45 * but WITHOUT ANY WARRANTY; without even the implied warranty of
46 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
47 * Lesser General Public License for more details.
48 *
49 * You should have received a copy of the GNU Lesser General Public
50 * License along with FFmpeg; if not, write to the Free Software
51 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
52 */
53
54 #include "libavutil/avassert.h"
55 #include "libavutil/avstring.h"
56 #include "libavutil/float_dsp.h"
57 #include "libavutil/intfloat.h"
58 #include "libavutil/mem.h"
59 #include "avcodec.h"
60 #include "bytestream.h"
61 #include "codec_internal.h"
62 #include "decode.h"
63 #include "get_bits.h"
64 #include "speexdata.h"
65
66 #define SPEEX_NB_MODES 3
67 #define SPEEX_INBAND_STEREO 9
68
69 #define QMF_ORDER 64
70 #define NB_ORDER 10
71 #define NB_FRAME_SIZE 160
72 #define NB_SUBMODES 9
73 #define NB_SUBMODE_BITS 4
74 #define SB_SUBMODE_BITS 3
75
76 #define NB_SUBFRAME_SIZE 40
77 #define NB_NB_SUBFRAMES 4
78 #define NB_PITCH_START 17
79 #define NB_PITCH_END 144
80
81 #define NB_DEC_BUFFER (NB_FRAME_SIZE + 2 * NB_PITCH_END + NB_SUBFRAME_SIZE + 12)
82
83 #define SPEEX_MEMSET(dst, c, n) (memset((dst), (c), (n) * sizeof(*(dst))))
84 #define SPEEX_COPY(dst, src, n) (memcpy((dst), (src), (n) * sizeof(*(dst))))
85
86 #define LSP_LINEAR(i) (.25f * (i) + .25f)
87 #define LSP_LINEAR_HIGH(i) (.3125f * (i) + .75f)
88 #define LSP_DIV_256(x) (0.00390625f * (x))
89 #define LSP_DIV_512(x) (0.001953125f * (x))
90 #define LSP_DIV_1024(x) (0.0009765625f * (x))
91
92 typedef struct LtpParams {
93 const int8_t *gain_cdbk;
94 int gain_bits;
95 int pitch_bits;
96 } LtpParam;
97
98 static const LtpParam ltp_params_vlbr = { gain_cdbk_lbr, 5, 0 };
99 static const LtpParam ltp_params_lbr = { gain_cdbk_lbr, 5, 7 };
100 static const LtpParam ltp_params_med = { gain_cdbk_lbr, 5, 7 };
101 static const LtpParam ltp_params_nb = { gain_cdbk_nb, 7, 7 };
102
103 typedef struct SplitCodebookParams {
104 int subvect_size;
105 int nb_subvect;
106 const signed char *shape_cb;
107 int shape_bits;
108 int have_sign;
109 } SplitCodebookParams;
110
111 static const SplitCodebookParams split_cb_nb_ulbr = { 20, 2, exc_20_32_table, 5, 0 };
112 static const SplitCodebookParams split_cb_nb_vlbr = { 10, 4, exc_10_16_table, 4, 0 };
113 static const SplitCodebookParams split_cb_nb_lbr = { 10, 4, exc_10_32_table, 5, 0 };
114 static const SplitCodebookParams split_cb_nb_med = { 8, 5, exc_8_128_table, 7, 0 };
115 static const SplitCodebookParams split_cb_nb = { 5, 8, exc_5_64_table, 6, 0 };
116 static const SplitCodebookParams split_cb_sb = { 5, 8, exc_5_256_table, 8, 0 };
117 static const SplitCodebookParams split_cb_high = { 8, 5, hexc_table, 7, 1 };
118 static const SplitCodebookParams split_cb_high_lbr= { 10, 4, hexc_10_32_table,5, 0 };
119
120 /** Quantizes LSPs */
121 typedef void (*lsp_quant_func)(float *, float *, int, GetBitContext *);
122
123 /** Decodes quantized LSPs */
124 typedef void (*lsp_unquant_func)(float *, int, GetBitContext *);
125
126 /** Long-term predictor quantization */
127 typedef int (*ltp_quant_func)(float *, float *, float *,
128 float *, float *, float *,
129 const void *, int, int, float, int, int,
130 GetBitContext *, char *, float *,
131 float *, int, int, int, float *);
132
133 /** Long-term un-quantize */
134 typedef void (*ltp_unquant_func)(float *, float *, int, int,
135 float, const void *, int, int *,
136 float *, GetBitContext *, int, int,
137 float, int);
138
139 /** Innovation quantization function */
140 typedef void (*innovation_quant_func)(float *, float *,
141 float *, float *, const void *,
142 int, int, float *, float *,
143 GetBitContext *, char *, int, int);
144
145 /** Innovation unquantization function */
146 typedef void (*innovation_unquant_func)(float *, const void *, int,
147 GetBitContext *, uint32_t *);
148
149 typedef struct SpeexSubmode {
150 int lbr_pitch; /**< Set to -1 for "normal" modes, otherwise encode pitch using
151 a global pitch and allowing a +- lbr_pitch variation (for
152 low not-rates)*/
153 int forced_pitch_gain; /**< Use the same (forced) pitch gain for all
154 sub-frames */
155 int have_subframe_gain; /**< Number of bits to use as sub-frame innovation
156 gain */
157 int double_codebook; /**< Apply innovation quantization twice for higher
158 quality (and higher bit-rate)*/
159 lsp_unquant_func lsp_unquant; /**< LSP unquantization function */
160
161 ltp_unquant_func ltp_unquant; /**< Long-term predictor (pitch) un-quantizer */
162 const void *LtpParam; /**< Pitch parameters (options) */
163
164 innovation_unquant_func innovation_unquant; /**< Innovation un-quantization */
165 const void *innovation_params; /**< Innovation quantization parameters*/
166
167 float comb_gain; /**< Gain of enhancer comb filter */
168 } SpeexSubmode;
169
170 typedef struct SpeexMode {
171 int modeID; /**< ID of the mode */
172 int (*decode)(AVCodecContext *avctx, void *dec, GetBitContext *gb, float *out);
173 int frame_size; /**< Size of frames used for decoding */
174 int subframe_size; /**< Size of sub-frames used for decoding */
175 int lpc_size; /**< Order of LPC filter */
176 float folding_gain; /**< Folding gain */
177 const SpeexSubmode *submodes[NB_SUBMODES]; /**< Sub-mode data for the mode */
178 int default_submode; /**< Default sub-mode to use when decoding */
179 } SpeexMode;
180
181 typedef struct DecoderState {
182 const SpeexMode *mode;
183 int modeID; /**< ID of the decoder mode */
184 int first; /**< Is first frame */
185 int full_frame_size; /**< Length of full-band frames */
186 int is_wideband; /**< If wideband is present */
187 int count_lost; /**< Was the last frame lost? */
188 int frame_size; /**< Length of high-band frames */
189 int subframe_size; /**< Length of high-band sub-frames */
190 int nb_subframes; /**< Number of high-band sub-frames */
191 int lpc_size; /**< Order of high-band LPC analysis */
192 float last_ol_gain; /**< Open-loop gain for previous frame */
193 float *innov_save; /**< If non-NULL, innovation is copied here */
194
195 /* This is used in packet loss concealment */
196 int last_pitch; /**< Pitch of last correctly decoded frame */
197 float last_pitch_gain; /**< Pitch gain of last correctly decoded frame */
198 uint32_t seed; /**< Seed used for random number generation */
199
200 int encode_submode;
201 const SpeexSubmode *const *submodes; /**< Sub-mode data */
202 int submodeID; /**< Activated sub-mode */
203 int lpc_enh_enabled; /**< 1 when LPC enhancer is on, 0 otherwise */
204
205 /* Vocoder data */
206 float voc_m1;
207 float voc_m2;
208 float voc_mean;
209 int voc_offset;
210
211 int dtx_enabled;
212 int highpass_enabled; /**< Is the input filter enabled */
213
214 float *exc; /**< Start of excitation frame */
215 float mem_hp[2]; /**< High-pass filter memory */
216 float exc_buf[NB_DEC_BUFFER]; /**< Excitation buffer */
217 float old_qlsp[NB_ORDER]; /**< Quantized LSPs for previous frame */
218 float interp_qlpc[NB_ORDER]; /**< Interpolated quantized LPCs */
219 float mem_sp[NB_ORDER]; /**< Filter memory for synthesis signal */
220 float g0_mem[QMF_ORDER];
221 float g1_mem[QMF_ORDER];
222 float pi_gain[NB_NB_SUBFRAMES]; /**< Gain of LPC filter at theta=pi (fe/2) */
223 float exc_rms[NB_NB_SUBFRAMES]; /**< RMS of excitation per subframe */
224 } DecoderState;
225
226 /* Default handler for user callbacks: skip it */
227 static int speex_default_user_handler(GetBitContext *gb, void *state, void *data)
228 {
229 const int req_size = get_bits(gb, 4);
230 skip_bits_long(gb, 5 + 8 * req_size);
231 return 0;
232 }
233
234 typedef struct StereoState {
235 float balance; /**< Left/right balance info */
236 float e_ratio; /**< Ratio of energies: E(left+right)/[E(left)+E(right)] */
237 float smooth_left; /**< Smoothed left channel gain */
238 float smooth_right; /**< Smoothed right channel gain */
239 } StereoState;
240
241 typedef struct SpeexContext {
242 AVClass *class;
243 GetBitContext gb;
244
245 int32_t version_id; /**< Version for Speex (for checking compatibility) */
246 int32_t rate; /**< Sampling rate used */
247 int32_t mode; /**< Mode used (0 for narrowband, 1 for wideband) */
248 int32_t bitstream_version; /**< Version ID of the bit-stream */
249 int32_t nb_channels; /**< Number of channels decoded */
250 int32_t bitrate; /**< Bit-rate used */
251 int32_t frame_size; /**< Size of frames */
252 int32_t vbr; /**< 1 for a VBR decoding, 0 otherwise */
253 int32_t frames_per_packet; /**< Number of frames stored per Ogg packet */
254 int32_t extra_headers; /**< Number of additional headers after the comments */
255
256 int pkt_size;
257
258 StereoState stereo;
259 DecoderState st[SPEEX_NB_MODES];
260
261 AVFloatDSPContext *fdsp;
262 } SpeexContext;
263
264 static void lsp_unquant_lbr(float *lsp, int order, GetBitContext *gb)
265 {
266 int id;
267
268 for (int i = 0; i < order; i++)
269 lsp[i] = LSP_LINEAR(i);
270
271 id = get_bits(gb, 6);
272 for (int i = 0; i < 10; i++)
273 lsp[i] += LSP_DIV_256(cdbk_nb[id * 10 + i]);
274
275 id = get_bits(gb, 6);
276 for (int i = 0; i < 5; i++)
277 lsp[i] += LSP_DIV_512(cdbk_nb_low1[id * 5 + i]);
278
279 id = get_bits(gb, 6);
280 for (int i = 0; i < 5; i++)
281 lsp[i + 5] += LSP_DIV_512(cdbk_nb_high1[id * 5 + i]);
282 }
283
284 static void forced_pitch_unquant(float *exc, float *exc_out, int start, int end,
285 float pitch_coef, const void *par, int nsf,
286 int *pitch_val, float *gain_val, GetBitContext *gb, int count_lost,
287 int subframe_offset, float last_pitch_gain, int cdbk_offset)
288 {
289 av_assert0(!isnan(pitch_coef));
290 pitch_coef = fminf(pitch_coef, .99f);
291 for (int i = 0; i < nsf; i++) {
292 exc_out[i] = exc[i - start] * pitch_coef;
293 exc[i] = exc_out[i];
294 }
295 pitch_val[0] = start;
296 gain_val[0] = gain_val[2] = 0.f;
297 gain_val[1] = pitch_coef;
298 }
299
300 static inline float speex_rand(float std, uint32_t *seed)
301 {
302 const uint32_t jflone = 0x3f800000;
303 const uint32_t jflmsk = 0x007fffff;
304 float fran;
305 uint32_t ran;
306 seed[0] = 1664525 * seed[0] + 1013904223;
307 ran = jflone | (jflmsk & seed[0]);
308 fran = av_int2float(ran);
309 fran -= 1.5f;
310 fran *= std;
311 return fran;
312 }
313
314 static void noise_codebook_unquant(float *exc, const void *par, int nsf,
315 GetBitContext *gb, uint32_t *seed)
316 {
317 for (int i = 0; i < nsf; i++)
318 exc[i] = speex_rand(1.f, seed);
319 }
320
321 24 static void split_cb_shape_sign_unquant(float *exc, const void *par, int nsf,
322 GetBitContext *gb, uint32_t *seed)
323 {
324 int subvect_size, nb_subvect, have_sign, shape_bits;
325 const SplitCodebookParams *params;
326 const signed char *shape_cb;
327 int signs[10], ind[10];
328
329 24 params = par;
330 24 subvect_size = params->subvect_size;
331 24 nb_subvect = params->nb_subvect;
332
333 24 shape_cb = params->shape_cb;
334 24 have_sign = params->have_sign;
335 24 shape_bits = params->shape_bits;
336
337 /* Decode codewords and gains */
338
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180 for (int i = 0; i < nb_subvect; i++) {
339
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156 signs[i] = have_sign ? get_bits1(gb) : 0;
340 156 ind[i] = get_bitsz(gb, shape_bits);
341 }
342 /* Compute decoded excitation */
343
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180 for (int i = 0; i < nb_subvect; i++) {
344
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156 const float s = signs[i] ? -1.f : 1.f;
345
346
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1116 for (int j = 0; j < subvect_size; j++)
347 960 exc[subvect_size * i + j] += s * 0.03125f * shape_cb[ind[i] * subvect_size + j];
348 }
349 24 }
350
351 #define SUBMODE(x) st->submodes[st->submodeID]->x
352
353 #define gain_3tap_to_1tap(g) (FFABS(g[1]) + (g[0] > 0.f ? g[0] : -.5f * g[0]) + (g[2] > 0.f ? g[2] : -.5f * g[2]))
354
355 static void
356 12 pitch_unquant_3tap(float *exc, float *exc_out, int start, int end, float pitch_coef,
357 const void *par, int nsf, int *pitch_val, float *gain_val, GetBitContext *gb,
358 int count_lost, int subframe_offset, float last_pitch_gain, int cdbk_offset)
359 {
360 int pitch, gain_index, gain_cdbk_size;
361 const int8_t *gain_cdbk;
362 const LtpParam *params;
363 float gain[3];
364
365 12 params = (const LtpParam *)par;
366 12 gain_cdbk_size = 1 << params->gain_bits;
367 12 gain_cdbk = params->gain_cdbk + 4 * gain_cdbk_size * cdbk_offset;
368
369 12 pitch = get_bitsz(gb, params->pitch_bits);
370 12 pitch += start;
371 12 gain_index = get_bitsz(gb, params->gain_bits);
372 12 gain[0] = 0.015625f * gain_cdbk[gain_index * 4] + .5f;
373 12 gain[1] = 0.015625f * gain_cdbk[gain_index * 4 + 1] + .5f;
374 12 gain[2] = 0.015625f * gain_cdbk[gain_index * 4 + 2] + .5f;
375
376
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12 if (count_lost && pitch > subframe_offset) {
377 float tmp = count_lost < 4 ? last_pitch_gain : 0.5f * last_pitch_gain;
378 float gain_sum;
379
380 tmp = fminf(tmp, .95f);
381 gain_sum = gain_3tap_to_1tap(gain);
382
383 if (gain_sum > tmp && gain_sum > 0.f) {
384 float fact = tmp / gain_sum;
385 for (int i = 0; i < 3; i++)
386 gain[i] *= fact;
387 }
388 }
389
390 12 pitch_val[0] = pitch;
391 12 gain_val[0] = gain[0];
392 12 gain_val[1] = gain[1];
393 12 gain_val[2] = gain[2];
394 12 SPEEX_MEMSET(exc_out, 0, nsf);
395
396
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48 for (int i = 0; i < 3; i++) {
397 int tmp1, tmp3;
398 36 int pp = pitch + 1 - i;
399 36 tmp1 = nsf;
400
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36 if (tmp1 > pp)
401 27 tmp1 = pp;
402
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990 for (int j = 0; j < tmp1; j++)
403 954 exc_out[j] += gain[2 - i] * exc[j - pp];
404 36 tmp3 = nsf;
405
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36 if (tmp3 > pp + pitch)
406 9 tmp3 = pp + pitch;
407
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468 for (int j = tmp1; j < tmp3; j++)
408 432 exc_out[j] += gain[2 - i] * exc[j - pp - pitch];
409 }
410 12 }
411
412 3 static void lsp_unquant_nb(float *lsp, int order, GetBitContext *gb)
413 {
414 int id;
415
416
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33 for (int i = 0; i < order; i++)
417 30 lsp[i] = LSP_LINEAR(i);
418
419 3 id = get_bits(gb, 6);
420
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33 for (int i = 0; i < 10; i++)
421 30 lsp[i] += LSP_DIV_256(cdbk_nb[id * 10 + i]);
422
423 3 id = get_bits(gb, 6);
424
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18 for (int i = 0; i < 5; i++)
425 15 lsp[i] += LSP_DIV_512(cdbk_nb_low1[id * 5 + i]);
426
427 3 id = get_bits(gb, 6);
428
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18 for (int i = 0; i < 5; i++)
429 15 lsp[i] += LSP_DIV_1024(cdbk_nb_low2[id * 5 + i]);
430
431 3 id = get_bits(gb, 6);
432
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18 for (int i = 0; i < 5; i++)
433 15 lsp[i + 5] += LSP_DIV_512(cdbk_nb_high1[id * 5 + i]);
434
435 3 id = get_bits(gb, 6);
436
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18 for (int i = 0; i < 5; i++)
437 15 lsp[i + 5] += LSP_DIV_1024(cdbk_nb_high2[id * 5 + i]);
438 3 }
439
440 6 static void lsp_unquant_high(float *lsp, int order, GetBitContext *gb)
441 {
442 int id;
443
444
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54 for (int i = 0; i < order; i++)
445 48 lsp[i] = LSP_LINEAR_HIGH(i);
446
447 6 id = get_bits(gb, 6);
448
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54 for (int i = 0; i < order; i++)
449 48 lsp[i] += LSP_DIV_256(high_lsp_cdbk[id * order + i]);
450
451 6 id = get_bits(gb, 6);
452
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54 for (int i = 0; i < order; i++)
453 48 lsp[i] += LSP_DIV_512(high_lsp_cdbk2[id * order + i]);
454 6 }
455
456 /* 2150 bps "vocoder-like" mode for comfort noise */
457 static const SpeexSubmode nb_submode1 = {
458 0, 1, 0, 0, lsp_unquant_lbr, forced_pitch_unquant, NULL,
459 noise_codebook_unquant, NULL, -1.f
460 };
461
462 /* 5.95 kbps very low bit-rate mode */
463 static const SpeexSubmode nb_submode2 = {
464 0, 0, 0, 0, lsp_unquant_lbr, pitch_unquant_3tap, &ltp_params_vlbr,
465 split_cb_shape_sign_unquant, &split_cb_nb_vlbr, .6f
466 };
467
468 /* 8 kbps low bit-rate mode */
469 static const SpeexSubmode nb_submode3 = {
470 -1, 0, 1, 0, lsp_unquant_lbr, pitch_unquant_3tap, &ltp_params_lbr,
471 split_cb_shape_sign_unquant, &split_cb_nb_lbr, .55f
472 };
473
474 /* 11 kbps medium bit-rate mode */
475 static const SpeexSubmode nb_submode4 = {
476 -1, 0, 1, 0, lsp_unquant_lbr, pitch_unquant_3tap, &ltp_params_med,
477 split_cb_shape_sign_unquant, &split_cb_nb_med, .45f
478 };
479
480 /* 15 kbps high bit-rate mode */
481 static const SpeexSubmode nb_submode5 = {
482 -1, 0, 3, 0, lsp_unquant_nb, pitch_unquant_3tap, &ltp_params_nb,
483 split_cb_shape_sign_unquant, &split_cb_nb, .25f
484 };
485
486 /* 18.2 high bit-rate mode */
487 static const SpeexSubmode nb_submode6 = {
488 -1, 0, 3, 0, lsp_unquant_nb, pitch_unquant_3tap, &ltp_params_nb,
489 split_cb_shape_sign_unquant, &split_cb_sb, .15f
490 };
491
492 /* 24.6 kbps high bit-rate mode */
493 static const SpeexSubmode nb_submode7 = {
494 -1, 0, 3, 1, lsp_unquant_nb, pitch_unquant_3tap, &ltp_params_nb,
495 split_cb_shape_sign_unquant, &split_cb_nb, 0.05f
496 };
497
498 /* 3.95 kbps very low bit-rate mode */
499 static const SpeexSubmode nb_submode8 = {
500 0, 1, 0, 0, lsp_unquant_lbr, forced_pitch_unquant, NULL,
501 split_cb_shape_sign_unquant, &split_cb_nb_ulbr, .5f
502 };
503
504 static const SpeexSubmode wb_submode1 = {
505 0, 0, 1, 0, lsp_unquant_high, NULL, NULL,
506 NULL, NULL, -1.f
507 };
508
509 static const SpeexSubmode wb_submode2 = {
510 0, 0, 1, 0, lsp_unquant_high, NULL, NULL,
511 split_cb_shape_sign_unquant, &split_cb_high_lbr, -1.f
512 };
513
514 static const SpeexSubmode wb_submode3 = {
515 0, 0, 1, 0, lsp_unquant_high, NULL, NULL,
516 split_cb_shape_sign_unquant, &split_cb_high, -1.f
517 };
518
519 static const SpeexSubmode wb_submode4 = {
520 0, 0, 1, 1, lsp_unquant_high, NULL, NULL,
521 split_cb_shape_sign_unquant, &split_cb_high, -1.f
522 };
523
524 static int nb_decode(AVCodecContext *, void *, GetBitContext *, float *);
525 static int sb_decode(AVCodecContext *, void *, GetBitContext *, float *);
526
527 static const SpeexMode speex_modes[SPEEX_NB_MODES] = {
528 {
529 .modeID = 0,
530 .decode = nb_decode,
531 .frame_size = NB_FRAME_SIZE,
532 .subframe_size = NB_SUBFRAME_SIZE,
533 .lpc_size = NB_ORDER,
534 .submodes = {
535 NULL, &nb_submode1, &nb_submode2, &nb_submode3, &nb_submode4,
536 &nb_submode5, &nb_submode6, &nb_submode7, &nb_submode8
537 },
538 .default_submode = 5,
539 },
540 {
541 .modeID = 1,
542 .decode = sb_decode,
543 .frame_size = NB_FRAME_SIZE,
544 .subframe_size = NB_SUBFRAME_SIZE,
545 .lpc_size = 8,
546 .folding_gain = 0.9f,
547 .submodes = {
548 NULL, &wb_submode1, &wb_submode2, &wb_submode3, &wb_submode4
549 },
550 .default_submode = 3,
551 },
552 {
553 .modeID = 2,
554 .decode = sb_decode,
555 .frame_size = 320,
556 .subframe_size = 80,
557 .lpc_size = 8,
558 .folding_gain = 0.7f,
559 .submodes = {
560 NULL, &wb_submode1
561 },
562 .default_submode = 1,
563 },
564 };
565
566 48 static float compute_rms(const float *x, int len)
567 {
568 48 float sum = 0.f;
569
570
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2928 for (int i = 0; i < len; i++)
571 2880 sum += x[i] * x[i];
572
573
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48 av_assert0(len > 0);
574 48 return sqrtf(.1f + sum / len);
575 }
576
577 static void bw_lpc(float gamma, const float *lpc_in,
578 float *lpc_out, int order)
579 {
580 float tmp = gamma;
581
582 for (int i = 0; i < order; i++) {
583 lpc_out[i] = tmp * lpc_in[i];
584 tmp *= gamma;
585 }
586 }
587
588 36 static void iir_mem(const float *x, const float *den,
589 float *y, int N, int ord, float *mem)
590 {
591
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1956 for (int i = 0; i < N; i++) {
592 1920 float yi = x[i] + mem[0];
593 1920 float nyi = -yi;
594
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16320 for (int j = 0; j < ord - 1; j++)
595 14400 mem[j] = mem[j + 1] + den[j] * nyi;
596 1920 mem[ord - 1] = den[ord - 1] * nyi;
597 1920 y[i] = yi;
598 }
599 36 }
600
601 3 static void highpass(const float *x, float *y, int len, float *mem, int wide)
602 {
603 static const float Pcoef[2][3] = {{ 1.00000f, -1.92683f, 0.93071f }, { 1.00000f, -1.97226f, 0.97332f } };
604 static const float Zcoef[2][3] = {{ 0.96446f, -1.92879f, 0.96446f }, { 0.98645f, -1.97277f, 0.98645f } };
605 const float *den, *num;
606
607 3 den = Pcoef[wide];
608 3 num = Zcoef[wide];
609
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483 for (int i = 0; i < len; i++) {
610 480 float yi = num[0] * x[i] + mem[0];
611 480 mem[0] = mem[1] + num[1] * x[i] + -den[1] * yi;
612 480 mem[1] = num[2] * x[i] + -den[2] * yi;
613 480 y[i] = yi;
614 }
615 3 }
616
617 #define median3(a, b, c) \
618 ((a) < (b) ? ((b) < (c) ? (b) : ((a) < (c) ? (c) : (a))) \
619 : ((c) < (b) ? (b) : ((c) < (a) ? (c) : (a))))
620
621 static int speex_std_stereo(GetBitContext *gb, void *state, void *data)
622 {
623 StereoState *stereo = data;
624 float sign = get_bits1(gb) ? -1.f : 1.f;
625
626 stereo->balance = exp(sign * .25f * get_bits(gb, 5));
627 stereo->e_ratio = e_ratio_quant[get_bits(gb, 2)];
628
629 return 0;
630 }
631
632 static int speex_inband_handler(GetBitContext *gb, void *state, StereoState *stereo)
633 {
634 int id = get_bits(gb, 4);
635
636 if (id == SPEEX_INBAND_STEREO) {
637 return speex_std_stereo(gb, state, stereo);
638 } else {
639 int adv;
640
641 if (id < 2)
642 adv = 1;
643 else if (id < 8)
644 adv = 4;
645 else if (id < 10)
646 adv = 8;
647 else if (id < 12)
648 adv = 16;
649 else if (id < 14)
650 adv = 32;
651 else
652 adv = 64;
653 skip_bits_long(gb, adv);
654 }
655 return 0;
656 }
657
658 12 static void sanitize_values(float *vec, float min_val, float max_val, int len)
659 {
660
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492 for (int i = 0; i < len; i++) {
661
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480 if (!isnormal(vec[i]) || fabsf(vec[i]) < 1e-8f)
662 120 vec[i] = 0.f;
663 else
664 360 vec[i] = av_clipf(vec[i], min_val, max_val);
665 }
666 12 }
667
668 24 static void signal_mul(const float *x, float *y, float scale, int len)
669 {
670
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984 for (int i = 0; i < len; i++)
671 960 y[i] = scale * x[i];
672 24 }
673
674 114 static float inner_prod(const float *x, const float *y, int len)
675 {
676 114 float sum = 0.f;
677
678
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1254 for (int i = 0; i < len; i += 8) {
679 1140 float part = 0.f;
680 1140 part += x[i + 0] * y[i + 0];
681 1140 part += x[i + 1] * y[i + 1];
682 1140 part += x[i + 2] * y[i + 2];
683 1140 part += x[i + 3] * y[i + 3];
684 1140 part += x[i + 4] * y[i + 4];
685 1140 part += x[i + 5] * y[i + 5];
686 1140 part += x[i + 6] * y[i + 6];
687 1140 part += x[i + 7] * y[i + 7];
688 1140 sum += part;
689 }
690
691 114 return sum;
692 }
693
694 12 static int interp_pitch(const float *exc, float *interp, int pitch, int len)
695 {
696 float corr[4][7], maxcorr;
697 int maxi, maxj;
698
699
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96 for (int i = 0; i < 7; i++)
700 84 corr[0][i] = inner_prod(exc, exc - pitch - 3 + i, len);
701
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48 for (int i = 0; i < 3; i++) {
702
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288 for (int j = 0; j < 7; j++) {
703 int i1, i2;
704 252 float tmp = 0.f;
705
706 252 i1 = 3 - j;
707
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252 if (i1 < 0)
708 108 i1 = 0;
709 252 i2 = 10 - j;
710
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252 if (i2 > 7)
711 108 i2 = 7;
712
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1584 for (int k = i1; k < i2; k++)
713 1332 tmp += shift_filt[i][k] * corr[0][j + k - 3];
714 252 corr[i + 1][j] = tmp;
715 }
716 }
717 12 maxi = maxj = 0;
718 12 maxcorr = corr[0][0];
719
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60 for (int i = 0; i < 4; i++) {
720
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384 for (int j = 0; j < 7; j++) {
721
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336 if (corr[i][j] > maxcorr) {
722 12 maxcorr = corr[i][j];
723 12 maxi = i;
724 12 maxj = j;
725 }
726 }
727 }
728
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972 for (int i = 0; i < len; i++) {
729 960 float tmp = 0.f;
730
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960 if (maxi > 0.f) {
731
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3840 for (int k = 0; k < 7; k++)
732 3360 tmp += exc[i - (pitch - maxj + 3) + k - 3] * shift_filt[maxi - 1][k];
733 } else {
734 480 tmp = exc[i - (pitch - maxj + 3)];
735 }
736 960 interp[i] = tmp;
737 }
738 12 return pitch - maxj + 3;
739 }
740
741 6 static void multicomb(const float *exc, float *new_exc, float *ak, int p, int nsf,
742 int pitch, int max_pitch, float comb_gain)
743 {
744 float old_ener, new_ener;
745 float iexc0_mag, iexc1_mag, exc_mag;
746 float iexc[4 * NB_SUBFRAME_SIZE];
747 float corr0, corr1, gain0, gain1;
748 float pgain1, pgain2;
749 float c1, c2, g1, g2;
750 float ngain, gg1, gg2;
751 6 int corr_pitch = pitch;
752
753 6 interp_pitch(exc, iexc, corr_pitch, 80);
754
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6 if (corr_pitch > max_pitch)
755 6 interp_pitch(exc, iexc + nsf, 2 * corr_pitch, 80);
756 else
757 interp_pitch(exc, iexc + nsf, -corr_pitch, 80);
758
759 6 iexc0_mag = sqrtf(1000.f + inner_prod(iexc, iexc, nsf));
760 6 iexc1_mag = sqrtf(1000.f + inner_prod(iexc + nsf, iexc + nsf, nsf));
761 6 exc_mag = sqrtf(1.f + inner_prod(exc, exc, nsf));
762 6 corr0 = inner_prod(iexc, exc, nsf);
763 6 corr1 = inner_prod(iexc + nsf, exc, nsf);
764
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6 if (corr0 > iexc0_mag * exc_mag)
765 pgain1 = 1.f;
766 else
767 6 pgain1 = (corr0 / exc_mag) / iexc0_mag;
768
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6 if (corr1 > iexc1_mag * exc_mag)
769 pgain2 = 1.f;
770 else
771 6 pgain2 = (corr1 / exc_mag) / iexc1_mag;
772 6 gg1 = exc_mag / iexc0_mag;
773 6 gg2 = exc_mag / iexc1_mag;
774
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6 if (comb_gain > 0.f) {
775 6 c1 = .4f * comb_gain + .07f;
776 6 c2 = .5f + 1.72f * (c1 - .07f);
777 } else {
778 c1 = c2 = 0.f;
779 }
780 6 g1 = 1.f - c2 * pgain1 * pgain1;
781 6 g2 = 1.f - c2 * pgain2 * pgain2;
782 6 g1 = fmaxf(g1, c1);
783 6 g2 = fmaxf(g2, c1);
784 6 g1 = c1 / g1;
785 6 g2 = c1 / g2;
786
787
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6 if (corr_pitch > max_pitch) {
788 6 gain0 = .7f * g1 * gg1;
789 6 gain1 = .3f * g2 * gg2;
790 } else {
791 gain0 = .6f * g1 * gg1;
792 gain1 = .6f * g2 * gg2;
793 }
794
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486 for (int i = 0; i < nsf; i++)
795 480 new_exc[i] = exc[i] + (gain0 * iexc[i]) + (gain1 * iexc[i + nsf]);
796 6 new_ener = compute_rms(new_exc, nsf);
797 6 old_ener = compute_rms(exc, nsf);
798
799 6 old_ener = fmaxf(old_ener, 1.f);
800 6 new_ener = fmaxf(new_ener, 1.f);
801 6 old_ener = fminf(old_ener, new_ener);
802 6 ngain = old_ener / new_ener;
803
804
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486 for (int i = 0; i < nsf; i++)
805 480 new_exc[i] *= ngain;
806 6 }
807
808 36 static void lsp_interpolate(const float *old_lsp, const float *new_lsp,
809 float *lsp, int len, int subframe,
810 int nb_subframes, float margin)
811 {
812 36 const float tmp = (1.f + subframe) / nb_subframes;
813
814
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348 for (int i = 0; i < len; i++) {
815 312 lsp[i] = (1.f - tmp) * old_lsp[i] + tmp * new_lsp[i];
816 312 lsp[i] = av_clipf(lsp[i], margin, M_PI - margin);
817 }
818
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276 for (int i = 1; i < len - 1; i++) {
819 240 lsp[i] = fmaxf(lsp[i], lsp[i - 1] + margin);
820
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240 if (lsp[i] > lsp[i + 1] - margin)
821 lsp[i] = .5f * (lsp[i] + lsp[i + 1] - margin);
822 }
823 36 }
824
825 36 static void lsp_to_lpc(const float *freq, float *ak, int lpcrdr)
826 {
827 float xout1, xout2, xin1, xin2;
828 float *pw, *n0;
829 36 float Wp[4 * NB_ORDER + 2] = { 0 };
830 float x_freq[NB_ORDER];
831 36 const int m = lpcrdr >> 1;
832
833 36 pw = Wp;
834
835 36 xin1 = xin2 = 1.f;
836
837
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348 for (int i = 0; i < lpcrdr; i++)
838 312 x_freq[i] = -cosf(freq[i]);
839
840 /* reconstruct P(z) and Q(z) by cascading second order
841 * polynomials in form 1 - 2xz(-1) +z(-2), where x is the
842 * LSP coefficient
843 */
844
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384 for (int j = 0; j <= lpcrdr; j++) {
845 348 int i2 = 0;
846
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1872 for (int i = 0; i < m; i++, i2 += 2) {
847 1524 n0 = pw + (i * 4);
848 1524 xout1 = xin1 + 2.f * x_freq[i2 ] * n0[0] + n0[1];
849 1524 xout2 = xin2 + 2.f * x_freq[i2 + 1] * n0[2] + n0[3];
850 1524 n0[1] = n0[0];
851 1524 n0[3] = n0[2];
852 1524 n0[0] = xin1;
853 1524 n0[2] = xin2;
854 1524 xin1 = xout1;
855 1524 xin2 = xout2;
856 }
857 348 xout1 = xin1 + n0[4];
858 348 xout2 = xin2 - n0[5];
859
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348 if (j > 0)
860 312 ak[j - 1] = (xout1 + xout2) * 0.5f;
861 348 n0[4] = xin1;
862 348 n0[5] = xin2;
863
864 348 xin1 = 0.f;
865 348 xin2 = 0.f;
866 }
867 36 }
868
869 3 static int nb_decode(AVCodecContext *avctx, void *ptr_st,
870 GetBitContext *gb, float *out)
871 {
872 3 DecoderState *st = ptr_st;
873 3 float ol_gain = 0, ol_pitch_coef = 0, best_pitch_gain = 0, pitch_average = 0;
874 3 int m, pitch, wideband, ol_pitch = 0, best_pitch = 40;
875 3 SpeexContext *s = avctx->priv_data;
876 float innov[NB_SUBFRAME_SIZE];
877 float exc32[NB_SUBFRAME_SIZE];
878 float interp_qlsp[NB_ORDER];
879 float qlsp[NB_ORDER];
880 float ak[NB_ORDER];
881 3 float pitch_gain[3] = { 0 };
882
883 3 st->exc = st->exc_buf + 2 * NB_PITCH_END + NB_SUBFRAME_SIZE + 6;
884
885
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3 if (st->encode_submode) {
886 do { /* Search for next narrowband block (handle requests, skip wideband blocks) */
887
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3 if (get_bits_left(gb) < 5)
888 return AVERROR_INVALIDDATA;
889 3 wideband = get_bits1(gb);
890
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3 if (wideband) /* Skip wideband block (for compatibility) */ {
891 int submode, advance;
892
893 submode = get_bits(gb, SB_SUBMODE_BITS);
894 advance = wb_skip_table[submode];
895 advance -= SB_SUBMODE_BITS + 1;
896 if (advance < 0)
897 return AVERROR_INVALIDDATA;
898 skip_bits_long(gb, advance);
899
900 if (get_bits_left(gb) < 5)
901 return AVERROR_INVALIDDATA;
902 wideband = get_bits1(gb);
903 if (wideband) {
904 submode = get_bits(gb, SB_SUBMODE_BITS);
905 advance = wb_skip_table[submode];
906 advance -= SB_SUBMODE_BITS + 1;
907 if (advance < 0)
908 return AVERROR_INVALIDDATA;
909 skip_bits_long(gb, advance);
910 wideband = get_bits1(gb);
911 if (wideband) {
912 av_log(avctx, AV_LOG_ERROR, "more than two wideband layers found\n");
913 return AVERROR_INVALIDDATA;
914 }
915 }
916 }
917
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3 if (get_bits_left(gb) < 4)
918 return AVERROR_INVALIDDATA;
919 3 m = get_bits(gb, 4);
920
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3 if (m == 15) /* We found a terminator */ {
921 return AVERROR_INVALIDDATA;
922
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3 } else if (m == 14) /* Speex in-band request */ {
923 int ret = speex_inband_handler(gb, st, &s->stereo);
924 if (ret)
925 return ret;
926
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3 } else if (m == 13) /* User in-band request */ {
927 int ret = speex_default_user_handler(gb, st, NULL);
928 if (ret)
929 return ret;
930
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3 } else if (m > 8) /* Invalid mode */ {
931 return AVERROR_INVALIDDATA;
932 }
933
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3 } while (m > 8);
934
935 3 st->submodeID = m; /* Get the sub-mode that was used */
936 }
937
938 /* Shift all buffers by one frame */
939 3 memmove(st->exc_buf, st->exc_buf + NB_FRAME_SIZE, (2 * NB_PITCH_END + NB_SUBFRAME_SIZE + 12) * sizeof(float));
940
941 /* If null mode (no transmission), just set a couple things to zero */
942
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3 if (st->submodes[st->submodeID] == NULL) {
943 float lpc[NB_ORDER];
944 float innov_gain = 0.f;
945
946 bw_lpc(0.93f, st->interp_qlpc, lpc, NB_ORDER);
947 innov_gain = compute_rms(st->exc, NB_FRAME_SIZE);
948 for (int i = 0; i < NB_FRAME_SIZE; i++)
949 st->exc[i] = speex_rand(innov_gain, &st->seed);
950
951 /* Final signal synthesis from excitation */
952 iir_mem(st->exc, lpc, out, NB_FRAME_SIZE, NB_ORDER, st->mem_sp);
953 st->count_lost = 0;
954
955 return 0;
956 }
957
958 /* Unquantize LSPs */
959 3 SUBMODE(lsp_unquant)(qlsp, NB_ORDER, gb);
960
961 /* Damp memory if a frame was lost and the LSP changed too much */
962
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3 if (st->count_lost) {
963 float fact, lsp_dist = 0;
964
965 for (int i = 0; i < NB_ORDER; i++)
966 lsp_dist = lsp_dist + FFABS(st->old_qlsp[i] - qlsp[i]);
967 fact = .6f * exp(-.2f * lsp_dist);
968 for (int i = 0; i < NB_ORDER; i++)
969 st->mem_sp[i] = fact * st->mem_sp[i];
970 }
971
972 /* Handle first frame and lost-packet case */
973
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3 if (st->first || st->count_lost)
974 3 memcpy(st->old_qlsp, qlsp, sizeof(st->old_qlsp));
975
976 /* Get open-loop pitch estimation for low bit-rate pitch coding */
977
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3 if (SUBMODE(lbr_pitch) != -1)
978 ol_pitch = NB_PITCH_START + get_bits(gb, 7);
979
980
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3 if (SUBMODE(forced_pitch_gain))
981 ol_pitch_coef = 0.066667f * get_bits(gb, 4);
982
983 /* Get global excitation gain */
984 3 ol_gain = expf(get_bits(gb, 5) / 3.5f);
985
986
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3 if (st->submodeID == 1)
987 st->dtx_enabled = get_bits(gb, 4) == 15;
988
989
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3 if (st->submodeID > 1)
990 3 st->dtx_enabled = 0;
991
992
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15 for (int sub = 0; sub < NB_NB_SUBFRAMES; sub++) { /* Loop on subframes */
993 12 float *exc, *innov_save = NULL, tmp, ener;
994 int pit_min, pit_max, offset, q_energy;
995
996 12 offset = NB_SUBFRAME_SIZE * sub; /* Offset relative to start of frame */
997 12 exc = st->exc + offset; /* Excitation */
998
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12 if (st->innov_save) /* Original signal */
999 12 innov_save = st->innov_save + offset;
1000
1001 12 SPEEX_MEMSET(exc, 0, NB_SUBFRAME_SIZE); /* Reset excitation */
1002
1003 /* Adaptive codebook contribution */
1004
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12 av_assert0(SUBMODE(ltp_unquant));
1005 /* Handle pitch constraints if any */
1006
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12 if (SUBMODE(lbr_pitch) != -1) {
1007 int margin = SUBMODE(lbr_pitch);
1008
1009 if (margin) {
1010 pit_min = ol_pitch - margin + 1;
1011 pit_min = FFMAX(pit_min, NB_PITCH_START);
1012 pit_max = ol_pitch + margin;
1013 pit_max = FFMIN(pit_max, NB_PITCH_START);
1014 } else {
1015 pit_min = pit_max = ol_pitch;
1016 }
1017 } else {
1018 12 pit_min = NB_PITCH_START;
1019 12 pit_max = NB_PITCH_END;
1020 }
1021
1022 12 SUBMODE(ltp_unquant)(exc, exc32, pit_min, pit_max, ol_pitch_coef, SUBMODE(LtpParam),
1023 NB_SUBFRAME_SIZE, &pitch, pitch_gain, gb, st->count_lost, offset,
1024 st->last_pitch_gain, 0);
1025
1026 12 sanitize_values(exc32, -32000, 32000, NB_SUBFRAME_SIZE);
1027
1028
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12 tmp = gain_3tap_to_1tap(pitch_gain);
1029
1030 12 pitch_average += tmp;
1031
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12 if ((tmp > best_pitch_gain &&
1032
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6 FFABS(2 * best_pitch - pitch) >= 3 &&
1033
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3 FFABS(3 * best_pitch - pitch) >= 4 &&
1034
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3 FFABS(4 * best_pitch - pitch) >= 5) ||
1035
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9 (tmp > .6f * best_pitch_gain &&
1036
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3 (FFABS(best_pitch - 2 * pitch) < 3 ||
1037
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3 FFABS(best_pitch - 3 * pitch) < 4 ||
1038
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3 FFABS(best_pitch - 4 * pitch) < 5)) ||
1039
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9 ((.67f * tmp) > best_pitch_gain &&
1040
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3 (FFABS(2 * best_pitch - pitch) < 3 ||
1041 FFABS(3 * best_pitch - pitch) < 4 ||
1042 FFABS(4 * best_pitch - pitch) < 5))) {
1043 6 best_pitch = pitch;
1044
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6 if (tmp > best_pitch_gain)
1045 6 best_pitch_gain = tmp;
1046 }
1047
1048 12 memset(innov, 0, sizeof(innov));
1049
1050 /* Decode sub-frame gain correction */
1051
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12 if (SUBMODE(have_subframe_gain) == 3) {
1052 12 q_energy = get_bits(gb, 3);
1053 12 ener = exc_gain_quant_scal3[q_energy] * ol_gain;
1054 } else if (SUBMODE(have_subframe_gain) == 1) {
1055 q_energy = get_bits1(gb);
1056 ener = exc_gain_quant_scal1[q_energy] * ol_gain;
1057 } else {
1058 ener = ol_gain;
1059 }
1060
1061
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12 av_assert0(SUBMODE(innovation_unquant));
1062 /* Fixed codebook contribution */
1063 12 SUBMODE(innovation_unquant)(innov, SUBMODE(innovation_params), NB_SUBFRAME_SIZE, gb, &st->seed);
1064 /* De-normalize innovation and update excitation */
1065
1066 12 signal_mul(innov, innov, ener, NB_SUBFRAME_SIZE);
1067
1068 /* Decode second codebook (only for some modes) */
1069
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12 if (SUBMODE(double_codebook)) {
1070 float innov2[NB_SUBFRAME_SIZE] = { 0 };
1071
1072 SUBMODE(innovation_unquant)(innov2, SUBMODE(innovation_params), NB_SUBFRAME_SIZE, gb, &st->seed);
1073 signal_mul(innov2, innov2, 0.454545f * ener, NB_SUBFRAME_SIZE);
1074 for (int i = 0; i < NB_SUBFRAME_SIZE; i++)
1075 innov[i] += innov2[i];
1076 }
1077
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492 for (int i = 0; i < NB_SUBFRAME_SIZE; i++)
1078 480 exc[i] = exc32[i] + innov[i];
1079
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12 if (innov_save)
1080 12 memcpy(innov_save, innov, sizeof(innov));
1081
1082 /* Vocoder mode */
1083
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12 if (st->submodeID == 1) {
1084 float g = ol_pitch_coef;
1085
1086 g = av_clipf(1.5f * (g - .2f), 0.f, 1.f);
1087
1088 SPEEX_MEMSET(exc, 0, NB_SUBFRAME_SIZE);
1089 while (st->voc_offset < NB_SUBFRAME_SIZE) {
1090 if (st->voc_offset >= 0)
1091 exc[st->voc_offset] = sqrtf(2.f * ol_pitch) * (g * ol_gain);
1092 st->voc_offset += ol_pitch;
1093 }
1094 st->voc_offset -= NB_SUBFRAME_SIZE;
1095
1096 for (int i = 0; i < NB_SUBFRAME_SIZE; i++) {
1097 float exci = exc[i];
1098 exc[i] = (.7f * exc[i] + .3f * st->voc_m1) + ((1.f - .85f * g) * innov[i]) - .15f * g * st->voc_m2;
1099 st->voc_m1 = exci;
1100 st->voc_m2 = innov[i];
1101 st->voc_mean = .8f * st->voc_mean + .2f * exc[i];
1102 exc[i] -= st->voc_mean;
1103 }
1104 }
1105 }
1106
1107
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3 if (st->lpc_enh_enabled && SUBMODE(comb_gain) > 0 && !st->count_lost) {
1108 3 multicomb(st->exc - NB_SUBFRAME_SIZE, out, st->interp_qlpc, NB_ORDER,
1109 3 2 * NB_SUBFRAME_SIZE, best_pitch, 40, SUBMODE(comb_gain));
1110 3 multicomb(st->exc + NB_SUBFRAME_SIZE, out + 2 * NB_SUBFRAME_SIZE,
1111 3 st->interp_qlpc, NB_ORDER, 2 * NB_SUBFRAME_SIZE, best_pitch, 40,
1112 3 SUBMODE(comb_gain));
1113 } else {
1114 SPEEX_COPY(out, &st->exc[-NB_SUBFRAME_SIZE], NB_FRAME_SIZE);
1115 }
1116
1117 /* If the last packet was lost, re-scale the excitation to obtain the same
1118 * energy as encoded in ol_gain */
1119
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3 if (st->count_lost) {
1120 float exc_ener, gain;
1121
1122 exc_ener = compute_rms(st->exc, NB_FRAME_SIZE);
1123 av_assert0(exc_ener + 1.f > 0.f);
1124 gain = fminf(ol_gain / (exc_ener + 1.f), 2.f);
1125 for (int i = 0; i < NB_FRAME_SIZE; i++) {
1126 st->exc[i] *= gain;
1127 out[i] = st->exc[i - NB_SUBFRAME_SIZE];
1128 }
1129 }
1130
1131
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15 for (int sub = 0; sub < NB_NB_SUBFRAMES; sub++) { /* Loop on subframes */
1132 12 const int offset = NB_SUBFRAME_SIZE * sub; /* Offset relative to start of frame */
1133 12 float pi_g = 1.f, *sp = out + offset; /* Original signal */
1134
1135 12 lsp_interpolate(st->old_qlsp, qlsp, interp_qlsp, NB_ORDER, sub, NB_NB_SUBFRAMES, 0.002f);
1136 12 lsp_to_lpc(interp_qlsp, ak, NB_ORDER); /* Compute interpolated LPCs (unquantized) */
1137
1138
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72 for (int i = 0; i < NB_ORDER; i += 2) /* Compute analysis filter at w=pi */
1139 60 pi_g += ak[i + 1] - ak[i];
1140 12 st->pi_gain[sub] = pi_g;
1141 12 st->exc_rms[sub] = compute_rms(st->exc + offset, NB_SUBFRAME_SIZE);
1142
1143 12 iir_mem(sp, st->interp_qlpc, sp, NB_SUBFRAME_SIZE, NB_ORDER, st->mem_sp);
1144
1145 12 memcpy(st->interp_qlpc, ak, sizeof(st->interp_qlpc));
1146 }
1147
1148
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3 if (st->highpass_enabled)
1149 3 highpass(out, out, NB_FRAME_SIZE, st->mem_hp, st->is_wideband);
1150
1151 /* Store the LSPs for interpolation in the next frame */
1152 3 memcpy(st->old_qlsp, qlsp, sizeof(st->old_qlsp));
1153
1154 3 st->count_lost = 0;
1155 3 st->last_pitch = best_pitch;
1156 3 st->last_pitch_gain = .25f * pitch_average;
1157 3 st->last_ol_gain = ol_gain;
1158 3 st->first = 0;
1159
1160 3 return 0;
1161 }
1162
1163 6 static void qmf_synth(const float *x1, const float *x2, const float *a, float *y, int N, int M, float *mem1, float *mem2)
1164 {
1165 6 const int M2 = M >> 1, N2 = N >> 1;
1166 float xx1[352], xx2[352];
1167
1168
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1446 for (int i = 0; i < N2; i++)
1169 1440 xx1[i] = x1[N2-1-i];
1170
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198 for (int i = 0; i < M2; i++)
1171 192 xx1[N2+i] = mem1[2*i+1];
1172
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1446 for (int i = 0; i < N2; i++)
1173 1440 xx2[i] = x2[N2-1-i];
1174
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198 for (int i = 0; i < M2; i++)
1175 192 xx2[N2+i] = mem2[2*i+1];
1176
1177
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726 for (int i = 0; i < N2; i += 2) {
1178 float y0, y1, y2, y3;
1179 float x10, x20;
1180
1181 720 y0 = y1 = y2 = y3 = 0.f;
1182 720 x10 = xx1[N2-2-i];
1183 720 x20 = xx2[N2-2-i];
1184
1185
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12240 for (int j = 0; j < M2; j += 2) {
1186 float x11, x21;
1187 float a0, a1;
1188
1189 11520 a0 = a[2*j];
1190 11520 a1 = a[2*j+1];
1191 11520 x11 = xx1[N2-1+j-i];
1192 11520 x21 = xx2[N2-1+j-i];
1193
1194 11520 y0 += a0 * (x11-x21);
1195 11520 y1 += a1 * (x11+x21);
1196 11520 y2 += a0 * (x10-x20);
1197 11520 y3 += a1 * (x10+x20);
1198 11520 a0 = a[2*j+2];
1199 11520 a1 = a[2*j+3];
1200 11520 x10 = xx1[N2+j-i];
1201 11520 x20 = xx2[N2+j-i];
1202
1203 11520 y0 += a0 * (x10-x20);
1204 11520 y1 += a1 * (x10+x20);
1205 11520 y2 += a0 * (x11-x21);
1206 11520 y3 += a1 * (x11+x21);
1207 }
1208 720 y[2 * i ] = 2.f * y0;
1209 720 y[2 * i+1] = 2.f * y1;
1210 720 y[2 * i+2] = 2.f * y2;
1211 720 y[2 * i+3] = 2.f * y3;
1212 }
1213
1214
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198 for (int i = 0; i < M2; i++)
1215 192 mem1[2*i+1] = xx1[i];
1216
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198 for (int i = 0; i < M2; i++)
1217 192 mem2[2*i+1] = xx2[i];
1218 6 }
1219
1220 6 static int sb_decode(AVCodecContext *avctx, void *ptr_st,
1221 GetBitContext *gb, float *out)
1222 {
1223 6 SpeexContext *s = avctx->priv_data;
1224 6 DecoderState *st = ptr_st;
1225 float low_pi_gain[NB_NB_SUBFRAMES];
1226 float low_exc_rms[NB_NB_SUBFRAMES];
1227 float interp_qlsp[NB_ORDER];
1228 int ret, wideband;
1229 float *low_innov_alias;
1230 float qlsp[NB_ORDER];
1231 float ak[NB_ORDER];
1232 const SpeexMode *mode;
1233
1234 6 mode = st->mode;
1235
1236
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6 if (st->modeID > 0) {
1237 6 low_innov_alias = out + st->frame_size;
1238 6 s->st[st->modeID - 1].innov_save = low_innov_alias;
1239 6 ret = speex_modes[st->modeID - 1].decode(avctx, &s->st[st->modeID - 1], gb, out);
1240
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6 if (ret < 0)
1241 return ret;
1242 }
1243
1244
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6 if (st->encode_submode) { /* Check "wideband bit" */
1245
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6 if (get_bits_left(gb) > 0)
1246 6 wideband = show_bits1(gb);
1247 else
1248 wideband = 0;
1249
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6 if (wideband) { /* Regular wideband frame, read the submode */
1250 6 wideband = get_bits1(gb);
1251 6 st->submodeID = get_bits(gb, SB_SUBMODE_BITS);
1252 } else { /* Was a narrowband frame, set "null submode" */
1253 st->submodeID = 0;
1254 }
1255
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6 if (st->submodeID != 0 && st->submodes[st->submodeID] == NULL)
1256 return AVERROR_INVALIDDATA;
1257 }
1258
1259 /* If null mode (no transmission), just set a couple things to zero */
1260
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6 if (st->submodes[st->submodeID] == NULL) {
1261 for (int i = 0; i < st->frame_size; i++)
1262 out[st->frame_size + i] = 1e-15f;
1263
1264 st->first = 1;
1265
1266 /* Final signal synthesis from excitation */
1267 iir_mem(out + st->frame_size, st->interp_qlpc, out + st->frame_size, st->frame_size, st->lpc_size, st->mem_sp);
1268
1269 qmf_synth(out, out + st->frame_size, h0, out, st->full_frame_size, QMF_ORDER, st->g0_mem, st->g1_mem);
1270
1271 return 0;
1272 }
1273
1274 6 memcpy(low_pi_gain, s->st[st->modeID - 1].pi_gain, sizeof(low_pi_gain));
1275 6 memcpy(low_exc_rms, s->st[st->modeID - 1].exc_rms, sizeof(low_exc_rms));
1276
1277 6 SUBMODE(lsp_unquant)(qlsp, st->lpc_size, gb);
1278
1279
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6 if (st->first)
1280 6 memcpy(st->old_qlsp, qlsp, sizeof(st->old_qlsp));
1281
1282
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30 for (int sub = 0; sub < st->nb_subframes; sub++) {
1283 float filter_ratio, el, rl, rh;
1284 24 float *innov_save = NULL, *sp;
1285 float exc[80];
1286 int offset;
1287
1288 24 offset = st->subframe_size * sub;
1289 24 sp = out + st->frame_size + offset;
1290 /* Pointer for saving innovation */
1291
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24 if (st->innov_save) {
1292 12 innov_save = st->innov_save + 2 * offset;
1293 12 SPEEX_MEMSET(innov_save, 0, 2 * st->subframe_size);
1294 }
1295
1296
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24 av_assert0(st->nb_subframes > 0);
1297 24 lsp_interpolate(st->old_qlsp, qlsp, interp_qlsp, st->lpc_size, sub, st->nb_subframes, 0.05f);
1298 24 lsp_to_lpc(interp_qlsp, ak, st->lpc_size);
1299
1300 /* Calculate reponse ratio between the low and high filter in the middle
1301 of the band (4000 Hz) */
1302 24 st->pi_gain[sub] = 1.f;
1303 24 rh = 1.f;
1304
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120 for (int i = 0; i < st->lpc_size; i += 2) {
1305 96 rh += ak[i + 1] - ak[i];
1306 96 st->pi_gain[sub] += ak[i] + ak[i + 1];
1307 }
1308
1309 24 rl = low_pi_gain[sub];
1310 24 filter_ratio = (rl + .01f) / (rh + .01f);
1311
1312 24 SPEEX_MEMSET(exc, 0, st->subframe_size);
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24 if (!SUBMODE(innovation_unquant)) {
1314 12 const int x = get_bits(gb, 5);
1315 12 const float g = expf(.125f * (x - 10)) / filter_ratio;
1316
1317
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492 for (int i = 0; i < st->subframe_size; i += 2) {
1318 480 exc[i ] = mode->folding_gain * low_innov_alias[offset + i ] * g;
1319 480 exc[i + 1] = -mode->folding_gain * low_innov_alias[offset + i + 1] * g;
1320 }
1321 } else {
1322 float gc, scale;
1323
1324 12 el = low_exc_rms[sub];
1325 12 gc = 0.87360f * gc_quant_bound[get_bits(gb, 4)];
1326
1327
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12 if (st->subframe_size == 80)
1328 gc *= M_SQRT2;
1329
1330 12 scale = (gc * el) / filter_ratio;
1331 12 SUBMODE(innovation_unquant)
1332 12 (exc, SUBMODE(innovation_params), st->subframe_size,
1333 gb, &st->seed);
1334
1335 12 signal_mul(exc, exc, scale, st->subframe_size);
1336
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12 if (SUBMODE(double_codebook)) {
1337 float innov2[80];
1338
1339 SPEEX_MEMSET(innov2, 0, st->subframe_size);
1340 SUBMODE(innovation_unquant)(innov2, SUBMODE(innovation_params), st->subframe_size, gb, &st->seed);
1341 signal_mul(innov2, innov2, 0.4f * scale, st->subframe_size);
1342 for (int i = 0; i < st->subframe_size; i++)
1343 exc[i] += innov2[i];
1344 }
1345 }
1346
1347
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24 if (st->innov_save) {
1348
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492 for (int i = 0; i < st->subframe_size; i++)
1349 480 innov_save[2 * i] = exc[i];
1350 }
1351
1352 24 iir_mem(st->exc_buf, st->interp_qlpc, sp, st->subframe_size, st->lpc_size, st->mem_sp);
1353 24 memcpy(st->exc_buf, exc, sizeof(exc));
1354 24 memcpy(st->interp_qlpc, ak, sizeof(st->interp_qlpc));
1355 24 st->exc_rms[sub] = compute_rms(st->exc_buf, st->subframe_size);
1356 }
1357
1358 6 qmf_synth(out, out + st->frame_size, h0, out, st->full_frame_size, QMF_ORDER, st->g0_mem, st->g1_mem);
1359 6 memcpy(st->old_qlsp, qlsp, sizeof(st->old_qlsp));
1360
1361 6 st->first = 0;
1362
1363 6 return 0;
1364 }
1365
1366 9 static int decoder_init(SpeexContext *s, DecoderState *st, const SpeexMode *mode)
1367 {
1368 9 st->mode = mode;
1369 9 st->modeID = mode->modeID;
1370
1371 9 st->first = 1;
1372 9 st->encode_submode = 1;
1373 9 st->is_wideband = st->modeID > 0;
1374 9 st->innov_save = NULL;
1375
1376 9 st->submodes = mode->submodes;
1377 9 st->submodeID = mode->default_submode;
1378 9 st->subframe_size = mode->subframe_size;
1379 9 st->lpc_size = mode->lpc_size;
1380
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9 st->full_frame_size = (1 + (st->modeID > 0)) * mode->frame_size;
1381 9 st->nb_subframes = mode->frame_size / mode->subframe_size;
1382 9 st->frame_size = mode->frame_size;
1383
1384 9 st->lpc_enh_enabled = 1;
1385
1386 9 st->last_pitch = 40;
1387 9 st->count_lost = 0;
1388 9 st->seed = 1000;
1389 9 st->last_ol_gain = 0;
1390
1391 9 st->voc_m1 = st->voc_m2 = st->voc_mean = 0;
1392 9 st->voc_offset = 0;
1393 9 st->dtx_enabled = 0;
1394 9 st->highpass_enabled = mode->modeID == 0;
1395
1396 9 return 0;
1397 }
1398
1399 3 static int parse_speex_extradata(AVCodecContext *avctx,
1400 const uint8_t *extradata, int extradata_size)
1401 {
1402 3 SpeexContext *s = avctx->priv_data;
1403 3 const uint8_t *buf = av_strnstr(extradata, "Speex ", extradata_size);
1404
1405
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3 if (!buf)
1406 return AVERROR_INVALIDDATA;
1407
1408 3 buf += 28;
1409
1410 3 s->version_id = bytestream_get_le32(&buf);
1411 3 buf += 4;
1412 3 s->rate = bytestream_get_le32(&buf);
1413
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3 if (s->rate <= 0)
1414 return AVERROR_INVALIDDATA;
1415 3 s->mode = bytestream_get_le32(&buf);
1416
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3 if (s->mode < 0 || s->mode >= SPEEX_NB_MODES)
1417 return AVERROR_INVALIDDATA;
1418 3 s->bitstream_version = bytestream_get_le32(&buf);
1419
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3 if (s->bitstream_version != 4)
1420 return AVERROR_INVALIDDATA;
1421 3 s->nb_channels = bytestream_get_le32(&buf);
1422
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3 if (s->nb_channels <= 0 || s->nb_channels > 2)
1423 return AVERROR_INVALIDDATA;
1424 3 s->bitrate = bytestream_get_le32(&buf);
1425 3 s->frame_size = bytestream_get_le32(&buf);
1426
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6 if (s->frame_size < NB_FRAME_SIZE << (s->mode > 1) ||
1427
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3 s->frame_size > INT32_MAX >> (s->mode > 1))
1428 return AVERROR_INVALIDDATA;
1429 3 s->frame_size = FFMIN(s->frame_size << (s->mode > 1), NB_FRAME_SIZE << s->mode);
1430 3 s->vbr = bytestream_get_le32(&buf);
1431 3 s->frames_per_packet = bytestream_get_le32(&buf);
1432
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3 if (s->frames_per_packet <= 0 ||
1433
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3 s->frames_per_packet > 64 ||
1434
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3 s->frames_per_packet >= INT32_MAX / s->nb_channels / s->frame_size)
1435 return AVERROR_INVALIDDATA;
1436 3 s->extra_headers = bytestream_get_le32(&buf);
1437
1438 3 return 0;
1439 }
1440
1441 3 static av_cold int speex_decode_init(AVCodecContext *avctx)
1442 {
1443 3 SpeexContext *s = avctx->priv_data;
1444 int ret;
1445
1446 3 s->fdsp = avpriv_float_dsp_alloc(0);
1447
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3 if (!s->fdsp)
1448 return AVERROR(ENOMEM);
1449
1450
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3 if (avctx->extradata && avctx->extradata_size >= 80) {
1451 3 ret = parse_speex_extradata(avctx, avctx->extradata, avctx->extradata_size);
1452
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3 if (ret < 0)
1453 return ret;
1454 } else {
1455 s->rate = avctx->sample_rate;
1456 if (s->rate <= 0)
1457 return AVERROR_INVALIDDATA;
1458
1459 s->nb_channels = avctx->ch_layout.nb_channels;
1460 if (s->nb_channels <= 0 || s->nb_channels > 2)
1461 return AVERROR_INVALIDDATA;
1462
1463 switch (s->rate) {
1464 case 8000: s->mode = 0; break;
1465 case 16000: s->mode = 1; break;
1466 case 32000: s->mode = 2; break;
1467 default: s->mode = 2;
1468 }
1469
1470 s->frames_per_packet = 64;
1471 s->frame_size = NB_FRAME_SIZE << s->mode;
1472 }
1473
1474
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3 if (avctx->codec_tag == MKTAG('S', 'P', 'X', 'N')) {
1475 int quality;
1476
1477 if (!avctx->extradata || avctx->extradata && avctx->extradata_size < 47) {
1478 av_log(avctx, AV_LOG_ERROR, "Missing or invalid extradata.\n");
1479 return AVERROR_INVALIDDATA;
1480 }
1481
1482 quality = avctx->extradata[37];
1483 if (quality > 10) {
1484 av_log(avctx, AV_LOG_ERROR, "Unsupported quality mode %d.\n", quality);
1485 return AVERROR_PATCHWELCOME;
1486 }
1487
1488 s->pkt_size = ((const uint8_t[]){ 5, 10, 15, 20, 20, 28, 28, 38, 38, 46, 62 })[quality];
1489
1490 s->mode = 0;
1491 s->nb_channels = 1;
1492 s->rate = avctx->sample_rate;
1493 if (s->rate <= 0)
1494 return AVERROR_INVALIDDATA;
1495 s->frames_per_packet = 1;
1496 s->frame_size = NB_FRAME_SIZE;
1497 }
1498
1499
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3 if (s->bitrate > 0)
1500 3 avctx->bit_rate = s->bitrate;
1501 3 av_channel_layout_uninit(&avctx->ch_layout);
1502 3 avctx->ch_layout.order = AV_CHANNEL_ORDER_UNSPEC;
1503 3 avctx->ch_layout.nb_channels = s->nb_channels;
1504 3 avctx->sample_rate = s->rate;
1505 3 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
1506
1507
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12 for (int m = 0; m <= s->mode; m++) {
1508 9 ret = decoder_init(s, &s->st[m], &speex_modes[m]);
1509
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9 if (ret < 0)
1510 return ret;
1511 }
1512
1513 3 s->stereo.balance = 1.f;
1514 3 s->stereo.e_ratio = .5f;
1515 3 s->stereo.smooth_left = 1.f;
1516 3 s->stereo.smooth_right = 1.f;
1517
1518 3 return 0;
1519 }
1520
1521 static void speex_decode_stereo(float *data, int frame_size, StereoState *stereo)
1522 {
1523 float balance, e_left, e_right, e_ratio;
1524
1525 balance = stereo->balance;
1526 e_ratio = stereo->e_ratio;
1527
1528 /* These two are Q14, with max value just below 2. */
1529 e_right = 1.f / sqrtf(e_ratio * (1.f + balance));
1530 e_left = sqrtf(balance) * e_right;
1531
1532 for (int i = frame_size - 1; i >= 0; i--) {
1533 float tmp = data[i];
1534 stereo->smooth_left = stereo->smooth_left * 0.98f + e_left * 0.02f;
1535 stereo->smooth_right = stereo->smooth_right * 0.98f + e_right * 0.02f;
1536 data[2 * i ] = stereo->smooth_left * tmp;
1537 data[2 * i + 1] = stereo->smooth_right * tmp;
1538 }
1539 }
1540
1541 3 static int speex_decode_frame(AVCodecContext *avctx, AVFrame *frame,
1542 int *got_frame_ptr, AVPacket *avpkt)
1543 {
1544 3 SpeexContext *s = avctx->priv_data;
1545 3 int frames_per_packet = s->frames_per_packet;
1546 3 const float scale = 1.f / 32768.f;
1547 3 int buf_size = avpkt->size;
1548 float *dst;
1549 int ret;
1550
1551
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3 if (s->pkt_size && avpkt->size == 62)
1552 buf_size = s->pkt_size;
1553
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3 if ((ret = init_get_bits8(&s->gb, avpkt->data, buf_size)) < 0)
1554 return ret;
1555
1556 3 frame->nb_samples = FFALIGN(s->frame_size * frames_per_packet, 4);
1557
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3 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1558 return ret;
1559
1560 3 dst = (float *)frame->extended_data[0];
1561
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3 for (int i = 0; i < frames_per_packet; i++) {
1562 3 ret = speex_modes[s->mode].decode(avctx, &s->st[s->mode], &s->gb, dst + i * s->frame_size);
1563
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3 if (ret < 0)
1564 return ret;
1565
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3 if (avctx->ch_layout.nb_channels == 2)
1566 speex_decode_stereo(dst + i * s->frame_size, s->frame_size, &s->stereo);
1567
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3 if (get_bits_left(&s->gb) < 5 ||
1568 show_bits(&s->gb, 5) == 15) {
1569 3 frames_per_packet = i + 1;
1570 3 break;
1571 }
1572 }
1573
1574 3 dst = (float *)frame->extended_data[0];
1575 3 s->fdsp->vector_fmul_scalar(dst, dst, scale, frame->nb_samples * frame->ch_layout.nb_channels);
1576 3 frame->nb_samples = s->frame_size * frames_per_packet;
1577
1578 3 *got_frame_ptr = 1;
1579
1580 3 return (get_bits_count(&s->gb) + 7) >> 3;
1581 }
1582
1583 3 static av_cold int speex_decode_close(AVCodecContext *avctx)
1584 {
1585 3 SpeexContext *s = avctx->priv_data;
1586 3 av_freep(&s->fdsp);
1587 3 return 0;
1588 }
1589
1590 const FFCodec ff_speex_decoder = {
1591 .p.name = "speex",
1592 CODEC_LONG_NAME("Speex"),
1593 .p.type = AVMEDIA_TYPE_AUDIO,
1594 .p.id = AV_CODEC_ID_SPEEX,
1595 .init = speex_decode_init,
1596 FF_CODEC_DECODE_CB(speex_decode_frame),
1597 .close = speex_decode_close,
1598 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
1599 .priv_data_size = sizeof(SpeexContext),
1600 .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
1601 };
1602